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android / platform / external / chromium_org / third_party / webrtc / ed94bc8eccb61b3fdffe5af5cdf090c20aefe63b
commited94bc8eccb61b3fdffe5af5cdf090c20aefe63b[log] [tgz]
authorasapersson@webrtc.org <asapersson@webrtc.org>Mon Nov 03 14:40:38 2014 +0000
committerasapersson@webrtc.org <asapersson@webrtc.org>Mon Nov 03 14:40:38 2014 +0000
tree90b2576b90f798ec11fa0da9d93012c111bd3375
parent9b1042c9f6a2d10ea222d1c3b748e2b8d321008b [diff]
Add stats for video:
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames

BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/24959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
8 files changed
tree: 90b2576b90f798ec11fa0da9d93012c111bd3375
  1. base/
  2. build/
  3. common_audio/
  4. common_video/
  5. examples/
  6. libjingle/
  7. modules/
  8. overrides/
  9. p2p/
  10. sound/
  11. system_wrappers/
  12. test/
  13. tools/
  14. video/
  15. video_engine/
  16. voice_engine/
  17. .gitignore
  18. BUILD.gn
  19. call.h
  20. common.gyp
  21. common.h
  22. common_types.h
  23. config.cc
  24. config.h
  25. engine_configurations.h
  26. experiments.h
  27. frame_callback.h
  28. LICENSE
  29. LICENSE_THIRD_PARTY
  30. OWNERS
  31. PATENTS
  32. PRESUBMIT.py
  33. README.chromium
  34. rtc_unittests.isolate
  35. supplement.gypi
  36. transport.h
  37. typedefs.h
  38. video_decoder.h
  39. video_encoder.h
  40. video_engine_tests.isolate
  41. video_frame.h
  42. video_receive_stream.h
  43. video_renderer.h
  44. video_send_stream.h
  45. webrtc.gyp
  46. webrtc_examples.gyp
  47. webrtc_perf_tests.isolate
  48. webrtc_tests.gypi